SIP Trunk vs Hosted PBX: Differences, Costs and How to Choose
SIP trunk vs hosted PBX is not a choice between equals: a SIP trunk is only the carrier connection that replaces PRI or analog lines and needs a PBX, while a hosted PBX is the phone system plus trunking run by a provider.
- A SIP trunk is transport only, so buying one without a PBX (your own or hosted) leaves you with no way to route extensions, voicemail or IVRs.
- Size SIP trunk channels by peak concurrent calls, starting at about 1 channel per 3 to 4 users, then verify against call logs.
- Hosted PBX is billed per user, while on-prem PBX with SIP trunks is billed per channel plus the hidden cost of hardware and admin time.
- Under roughly 25 users with no IT staff, hosted PBX usually wins on effort and risk; above 50 users or with analog, paging or door hardware, on-prem PBX with SIP trunks often wins on both cost and fit.
- Either option fails without good LAN and upload bandwidth, QoS, SIP ALG disabled, correct E911 addresses and early number porting.
Is a SIP trunk the same as a hosted PBX?
No. In the SIP trunk vs hosted PBX question, the two sit at different layers. A SIP trunk is a carrier connection that carries calls between your phone system and the public phone network (PSTN). A hosted PBX (also sold as UCaaS or cloud PBX) is the phone system itself, with trunking bundled in and operated by the provider.
So the real question is two questions: who runs the PBX (you or a provider), and how does it reach the carrier (PRI, analog, SIP trunk, or bundled). A SIP trunk always needs a PBX on the other end, such as Asterisk/FreePBX, 3CX, or a cloud-hosted instance. Without one, you have phone numbers with nothing to answer them.
A PBX handles extensions, ring groups, IVR, voicemail, call recording and desk phone registration. A trunk does none of that. It delivers an INVITE for a number you own and accepts authenticated outbound calls, by registration or by IP address.
PRI (Primary Rate Interface) is the older circuit a SIP trunk usually replaces. In North America a PRI is a T1 with 23 voice channels at 64 kbps plus 1 signaling channel; in Europe an E1 gives 30. The count is fixed, so a T1 gives exactly 23 calls whether you need 4 or 23.
Many carriers are retiring copper and TDM services, with higher prices or end-of-life dates, so confirm your own carrier's schedule. A SIP trunk lets you buy 6 channels instead of 23 and add more in minutes. The trade-off is that call quality now depends on your internet connection.
Who manages what, and what happens in an internet outage?
Responsibility is the practical difference, more than technology. The first table shows the usual split; contracts vary, so confirm which side owns each item.
Even with hosted PBX, the LAN, Wi-Fi, switches, firewall and internet link are still yours. Most bad hosted voice tickets trace back to these, not to the provider.
| Item | On-prem PBX + PRI | On-prem PBX + SIP trunk | Hosted PBX |
|---|---|---|---|
| PBX software, patching, backups | You | You | Provider |
| Dial plan, IVR, ring groups | You | You | You in a portal, within platform limits |
| Carrier connection | Carrier circuit plus your PRI card or gateway | Separate ITSP over internet or private circuit | Provider (bundled) |
| Toll-fraud and security | You | You, plus trunk limits at the ITSP | Shared: provider secures the platform, you secure users and passwords |
| E911 addresses | Carrier and you | You register and keep current | You enter per user or location |
| Internet link outage | Calls survive on PRI if PBX is up | Trunk calls fail without failover; internal calls continue | External and often internal calls fail |
How much does each option cost for 10, 25 and 100 users?
The figures below are illustrative monthly estimates using assumed prices, not quotes. Admin time is valued at $60 per hour.
At 10 users on-prem is about $110/month cheaper on these assumptions, and that gap depends mostly on the admin-hour assumption. The table assumes only 1 admin hour a month, and one failure or missed patch can use up the saving.
The break-even point shows how sensitive this is. At 10 users, non-admin on-prem costs are about $50/month against $220 hosted, so on-prem stops being cheaper at roughly 2.8 admin hours a month. At 25 users the break-even is about 7.7 hours ($85 against $550), and at 100 users about 28 hours ($310 against $2,000).
Add phones, licensing if you use 3CX or a commercial PBX, and one-off setup. Hosted pricing may also bundle call recording, mobile apps or video that you would otherwise pay for.
| Users | Hosted PBX | On-prem PBX + SIP trunk | PRI (23 channels, reference) |
|---|---|---|---|
| 10 | 10 x $22 = $220 | 4 channels ($20) + 10 DIDs ($10) + server ($20) + 1 admin hour ($60) = about $110 | About $400 plus gateway; fixed regardless of use |
| 25 | 25 x $22 = $550 | 8 channels ($40) + 25 DIDs ($25) + server ($20) + 3 admin hours ($180) = about $265 | Same $400, far more capacity than needed |
| 100 | 100 x $20 = $2,000 | 30 channels ($150) + 100 DIDs ($100) + redundant servers ($60) + 10 admin hours ($600) = about $910 | Needs multiple PRIs; usually uneconomic |
How many SIP trunk channels do I need?
Size channels by peak concurrent calls, not headcount. A common starting point is 1 channel per 3 to 4 users, so 25 users start at 7 to 8 channels. A call center may need 1 per 1 to 2 users, while a quiet office may need 1 per 6.
Check real data. Count overlapping calls in your PBX or PRI logs, take the busiest hour of a typical month, then add about 20 percent headroom. A call forwarded to a mobile uses two channels, one in and one out.
Bandwidth follows the same count. G.711 uses roughly 80 to 100 kbps per call each way including IP overhead, so 8 concurrent calls need about 800 kbps up and down. G.729 cuts that to roughly 30 kbps but may need a transcoding license and lowers audio quality.
Hosted PBX hides channel counts: you pay per seat and the provider handles capacity. Your internet link still has to carry the peak call count.
SIP trunk vs hosted PBX: which should I choose?
On-prem PBX with SIP trunks fits when you own a PBX or analog hardware (paging, door phones, fax, alarm lines), need custom integrations, face data-control rules, have many users or shared trunking across sites, and have Asterisk, FreePBX or 3CX skills in-house.
Hosted PBX fits when you have under about 25 users, no IT staff, several small sites or remote workers, and want fast setup with predictable per-seat cost.
Add the points for your situation below and see which column scores higher.
| Factor | Points toward hosted PBX | Points toward on-prem PBX + SIP trunk |
|---|---|---|
| Users | Under 25: +2. 25 to 50: +1 | 50 to 100: +1. Over 100: +2 |
| IT skills | None or generalist: +2 | Telephony admin in-house: +2 |
| Existing equipment | Mostly new IP phones: +1 | Analog lines, paging, fax, door entry: +2 |
| Sites | Several small sites or remote staff: +2 | One large site or central trunking: +1 |
| Internet | Stable fiber or cable with spare upload: +1 | Weak or single link: +1 (internal calls survive) |
| Compliance and data control | Standard needs: +0 | Strict control or custom integrations: +2 |
What network setup do both options need?
The prerequisites are the same for either model. Voice is sensitive to delay, jitter and loss, not to raw speed. SIP signaling normally uses UDP/TCP 5060 or TLS 5061, and RTP media uses a UDP range often set to 10000 to 20000.
Steps to prepare the network:
- Put phones on a voice VLAN and mark RTP as DSCP EF (46) and SIP signaling as CS3 (24) on switches and the router.
- Disable SIP ALG on the router or firewall; it rewrites SIP headers and causes one-way audio and dropped registrations.
- Confirm upload bandwidth: about 100 kbps per concurrent G.711 call plus 25 to 30 percent headroom, and prioritize voice on the WAN.
- Use a static public IP or a registration-based trunk. IP-authenticated trunks need your static address allow-listed at the carrier.
- Open firewall rules only to the provider's signaling IPs on 5060/5061 and the RTP range, never to the world. An SBC or PBX firewall also hides topology and blocks scans.
- For Asterisk behind NAT, set external_media_address, external_signaling_address and local_net on the PJSIP transport, as in the example below.
; /etc/asterisk/pjsip.conf (excerpt)
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
external_media_address=203.0.113.10
external_signaling_address=203.0.113.10
local_net=192.168.1.0/24
; debug when audio fails
; asterisk -rx "pjsip set logger on"
; asterisk -rx "rtp set debug on"
; sngrepWhat are the common mistakes, and are hybrid options possible?
Common mistakes: buying SIP trunks without a PBX; sizing channels by user count; ignoring E911 registration; assuming hosted removes the need for a good LAN; signing multi-year lock-ins; and expecting proprietary phones from an old PBX to work on a new one.
Number porting needs a Letter of Authorization (LOA) and details matching your current carrier account exactly, and it often takes days to weeks. Do not cancel the old lines until the port completes and test calls work. Keep a temporary forward in place during cutover.
Hybrid options exist. You can pair a hosted PBX with an on-prem analog gateway (ATA/FXS) for fax, alarm and paging, or run Asterisk/FreePBX on a cloud VPS with SIP trunks for control without local hardware. For cutover steps, follow a PBX to hosted VoIP migration checklist, and check codec guides if quality or bandwidth is a concern.
Common questions
Do I need a PBX if I have a SIP trunk?
Yes. A SIP trunk only carries calls to and from the public network, so something must answer, route and manage them. That can be an on-prem PBX such as FreePBX or 3CX, a cloud-hosted PBX instance you run, or a hosted PBX that includes its own trunking.
Which is better for a small business, SIP trunking or hosted PBX?
For most small businesses with no IT staff and under about 25 users, hosted PBX is simpler, because the provider runs the PBX, patching and carrier link. SIP trunking with your own PBX is better when you already own telephony hardware, need integrations, or have the skills to maintain it.
What is the difference between a SIP trunk and PRI?
A PRI is a fixed-capacity TDM circuit (23 voice channels on a T1, 30 on an E1) that needs a card or gateway. A SIP trunk delivers the same voice calls over IP, with channels you can add or remove in small steps, but it depends on your network quality.
Is on-premises PBX with SIP trunks cheaper than a cloud PBX?
On monthly running cost, usually yes: in the cost table it is $110 vs $220 at 10 users, $265 vs $550 at 25 and $910 vs $2,000 at 100. The saving holds only while admin time stays below the break-even of about 2.8 hours a month at 10 users, 7.7 at 25 and 28 at 100 (at $60/hour). Hardware, licensing, backups and security work push you toward that limit, which is why small teams without IT staff often still pick hosted.
Can I keep my existing phone numbers?
Usually yes, through number porting. You need a Letter of Authorization and details exactly matching your current carrier account, and the process can take from days to several weeks. Keep the old service active until the port completes.